Revision as of 12:56, 9 August 2017

Contents

Summary

PRELIMINARY REPORT -- Tests for this product are still ongoing (last updated August 9th, 2017) and may (and probably will) change.

Remarks

To use SRTP and TLS you have to ask Vozelia to activate it. For Fax there is a seperate registration to another trunk.

List of Issues found in media-relay Configuration

180 RINGING

The provider does not send a 180 Ringing response when the called party alerts.

FAX AUDIO

The provider does not fully support Audiofax (i.e. non-T.38). The provider supports Fax

FAX T38

The provider does not fully support T.38 fax

FAX T38ANDAUDIO

The provider does not support fallback to audio-fax if T.38 fails.

HOLD RETRIEVE

The provider does not support Hold/Retrieve of a call.

Our tests of this feature have shown unstable results (that is, the feature sometimes worked and sometimes did not). This may for example be caused by different equipment used at the provider side (e.g. media gateways) which behave differently.

RALERT DISC

Call disconnected by far end during alert does not disconnect locally

REDIR 302

The provider does not support external call redirection using the SIP 302 Redirect response

Test Results

This SIP provider requires a media-relay configuration. That is, all media traffic between the SIP provider and all endpoints must flow through the SBC. For this reason, a configuration without media-relay has not been tested and hence no test results for this configuration are listed.

Configuration with media-relay

Registration

The provider supports UDP and TCP as transport protocol. The tests were completed using TCP, since UDP is an unreliable protocol and requires all involved network elements to support IP-fragmentation. The provider supports TLS as well. It has not been tested by us yet.

NAT Traversal

The provider detects clients behind NAT and can handle calls to them without requiring the clients to use NAT-traversal methods like STUN. However MediaRelay is required, since the provider does not support a change of the remote RTP-endpoint during a call.

DTMF (RFC2833)

The provider does not support conveying telephony events (a.k.a. DTMF) in-band as RTP payload. However, it supports conveying telephony events (DTMF) using the SIP-INFO method. innovaphone endpoints will work with this method too. 3rd party endpoints may have issues though. In some cases there were problems with dtmf tones on international calls. If so you can contact Vozelia and they fix it.

Session Timer

The tests regarding the SIP-session timer were successful. This means that either no session expiry is used or that it is used and works. It does not imply that session expiry actually is used.

Redundancy

Registration of two SIP-interfaces on the same SIP-account is supported by the provider. However, the provider has no failover mechanism if one device is down. As a result, you can use both SIP-interfaces for load-balancing purposes. If one device is down, for 2 minutes incoming and outgoing calls might be rejected/fail.

Correct signalling of Ringing-state

Ringing is not signalled by the provider. This will lead to incorrect call-state display on the PBX (phone-UI, myPBX, Soap) for outbound calls to the PSTN. The caller will see no status-update on the phone-display/PC-screen, showing that the remote party was reached and is ringing.

Additionally external callers forwarded/transferred back to the PSTN, may get no ring-tone but hear silence while the remote party is ringing. This silence while waiting might lead to aborting the call. As a result, Carrier w/o Alerting is required in all PBX 'Mobility' objects.

An outgoing call that is disconnected by the far end during alert is not disconnected locally. Typically, the provider may play a message to the effect that the call has been rejected or the call can not be completed so that the calling user will hang up. This may be OK, however, it may create issues with automated calls (e.g. fax or modem) which do not listen to the announcement and keep waiting for the far end to accept the call.

CLIR

OK

Clip No Screening (CLNS)

Straight clip no screening (i.e. sending a foreign number as calling line id) works fine. As this works anyway, it does not matter, if the provider supports the interpretation of History-Info: or Diversion: SIP headers for providing the correct calling line id for diverted calls.

COLP

Outbound and inbound calls to/from the PSTN show the correct connected number.

For outbound calls to the PSTN, an update of the connected number is not signalled to the caller.

Early-Media

The provider supports early-media for outbound calls (hence inbound early media) to the PSTN.

Fax

Audio-Fax calls (that is, fax calls without T.38) do not work. However, all fax endpoints must be configured with exclusive codec "G711A".

Transport of faxes using T.38 worked to onnet destinations. However it failed to PSTN destination and moreover fallback to audiofax failed also.

As a result, T.38 is disabled on the SIP-interface and the use of audio-fax is necessary.

Codecs

supported to/from PSTN: G711A and G729

supported onnet (VoIP to VoIP): G711A and G729

IP-Fragmentation

OK

Large SIP messages

OK

Reverse Media Negotiation

OK

Mobility Calls

OK

As the provider does not provide proper alert signalling, you will need to configure the Carrier w/o Alerting check-mark in the PBX Mobility object.

SRTP

The provider supports audio encryption using SRTP for incoming, outgoing and on-net calls.

Provider supports dialling subscriber numbers

OK

Call Transfer

OK

Configuration

Use profile ES-Vozelia-Telefonia_IP in Gateway/Interfaces/SIP to configure this SIP provider.

Disclaimer

These tests look at a number of interoperability scenarios between innovaphone SIP devices and a given SIP trunk product. As we are enhancing our testing procedures, nature and number of these tests will vary.

All test results document the fact how the tested combination performs in the tested scenario. It explicitly does not comment on the question what the reason is for the behaviour nor if and how it could be changed. It thus does not imply that either the SIP trunk provider or the innovaphone device fails in any way. It merely says that the combination does not perform as defined by the test.

If not mentioned otherwise, all tests are repeated on a weekly basis (a.k.a. nightly tests) using the then-current innovaphone firmware. If a tested combination keeps performing differently from the state documented here during nightly tests, we reserve the right to update this article accordingly.

Some tests do not have consistent results. This may occur for various reasons. If so, we document the tests result as being unstable.

Some of the tested SIP trunk products are not available for nightly tests. In this case, the fact will be noted in the Summary section.